1. Field of the Invention
This invention relates to a high-resolution digital filter of a type which includes a memory structure receiving as an input a sampled digital signal, and an adder chain, with delay blocks therebetween, which adders are connected to outputs of the memory in order to convert the input into an output signal having predetermined frequency response characteristics.
The invention also relates to a method of filtering a sampled digital code signal.
The invention is more particularly, though not exclusively, directed to digital symmetrical FIR (Finite Impulse Response) filters, and throughout the specification, reference will be made to such an application for convenience of illustration.
2. Discussion of the Related Art
As is known, digital filters are devices intended for converting a sampled signal, received as an input, into another sampled signal having predetermined frequency response characteristics. A sampled signal obviously includes a digital signal which is coded with a predetermined number n of bits on which the filter accuracy, or resolution, is dependent.
Digital filters are primarily used in digital oscilloscopes, spectrum analyzers, and audio and video signal processors. In addition, such filters are gaining increasing acceptance on account of a number of advantages that they afford over corresponding analog filters.
For the same function, in fact, digital filters allow very narrow transmission bands, and are more stable both over time and variations in the power supply and operating temperature.
Digital filters have been implemented within integrated circuits employing digital multipliers and adders.
Digital multipliers can be implemented with a non-volatile memory structure, such as a look-up table, wherein the products of the input signal samples with the coefficients of the filter transfer function are stored.
A structure of this kind is described, for example, in an article entitled "30-MSamples/s Programmable Filter Processor", IEEE Journal of Solid-State Circuits, Vol. 25, No. 6, December 1990, and in Italian Patent Application No. 22954-A/88 filed by this same Applicant which are herein incorporated by reference.
While being in many ways advantageous, this prior approach has a drawback as explained herein below.
When the number of bits used for sampling the input signal is denoted by n, the total number of the filter coefficients is denoted by N, and p denotes the number of bits required to store the product of the samples with the coefficients, then it is found that the memory size is given by 2 nNp. It is readily determined from the above that increasing the sampling of the input signal even by one bit only, to thereby enhance the filter resolution, would result in the memory size being doubled. By way of example, assume a hypothetical transition from 8-bits sampling, typical of current applications, to n=12 bits sampling, as would be highly desirable to improve filtering performance on audio and video signals.
A suitable, hypothetical memory structure would have to be a size sixteen times as large as that required by the former, 8-bits, coding. Such a memory would, therefore, occupy an inordinate amount of space on an integrated circuit. Moreover, it would lack adequate speed of data access because access time is heavily dependent on both the increased coding complexity and the memory size, specifically the number of bits per row. The state of the art offers no satisfactory solution to the problem of circumventing this vast memory expansion whenever improved filtering accuracy or resolution is sought.
The underlying technical problem of this invention is to provide a digital filter, and associated filtering method, having such structural and functional features as to enable high-resolution processing of digital sampled signals coded with a large number of bits, to overcome the limitations of the approaches currently proposed in the state of the art.
The idea on which this invention stands is one of splitting the sampled signal coding into at least two portions and then filtering each portion separately one from the other, and to ultimately re-construct the sampled output signal.